HFR (High Frequency Reconstruction) techniques, such as Spectral Band Replication (SBR), allow for a significant improvement of the coding efficiency of traditional perceptual audio codecs. In combination with MPEG-4 Advanced Audio Coding (AAC), HFR forms a very efficient audio codec, which is already in use within the XM Satellite Radio system and Digital Radio Mondiale, and also standardized within 3GPP, DVD Forum and others. The combination of AAC and SBR is called aacPlus. It is part of the MPEG-4 standard where it is referred to as the High Efficiency AAC Profile (HE-AAC). In general, HFR technologies can be combined with any perceptual audio codec in a back and forward compatible way, thus offering the possibility to upgrade already established broadcasting systems like the MPEG Layer-2 used in the Eureka DAB system. HFR transposition methods can also be combined with speech codecs to allow wide band speech at ultra low bit rates.
The basic idea behind HRF (or SBR in particular) is the observation that there usually exists a strong correlation between the characteristics of the high frequency range of a signal (referred to as the high frequency component) and the characteristics of the low frequency range of the same signal (referred to as the low frequency component). Thus, a good approximation for the representation of the original input high frequency range of a signal can be achieved by a signal transposition from the low frequency range to the high frequency range.
Audio signals may be provided at different sampling rates. Users of an audio codec typically want to be able to encode audio signals at various input sampling rates. In a similar manner, users of an audio codec want to be able to select various sampling rates at an output of the audio decoder. By way of example, a user makes use of an audio codec to encode uncompressed audio signals (e.g. from a compact disk, from way-files, or from media libraries). These uncompressed audio signals may be at various input sampling rates such as 24, 32, 44.1 or 48 kHz which are supported by various rendering devices (TV, mp3 players, smart phones, etc.).
As such, the audio codec should be able to handle various sampling rates at the input to the encoder and should be able to provide various sampling rates at the output of the decoder. In particular, the audio codec should be able to convert the sampling rates of audio signals at the input and at the output of the audio codec in a flexible and processor efficient manner. By way of example, a user may select an output sampling rate of 48 kHz vs. and input sampling rate of 24 kHz. In this case, the audio codec should be able to provide a sampling rate conversion (upsampling by a factor of two) which requires low computational complexity. In particular, the computational complexity related to the upsampling should be reduced (or, if possible, the necessity of explicit upsampling, using a conventional resampler, should be removed completely).
The present document describes audio codecs which make use of high frequency reconstruction, notably audio codecs using SBR, which are configured to perform sampling rate conversion of audio signals at reduced computational complexity.